For many years voice telephone service was implemented over a circuit switched network commonly known as the public switched telephone network (PSTN) and controlled by a local telephone service provider. In such systems, the analog electrical signals representing the conversation are transmitted between the two telephone handsets on a dedicated twisted-pair-copper-wire circuit. More specifically, each of the two endpoint telephones is coupled to a local switching station by a dedicated pair of copper wires known as a subscriber loop. The two switching stations are connected by a trunk line network comprising multiple copper wire pairs. When a telephone call is placed, the circuit is completed by dynamically coupling each subscriber loop to a dedicated pair of copper wires in the trunk line network that completes the circuit between the two local switching stations.
Because each call placed by, or to, a subscriber loop must route through the local switching station, billing for calling services may be readily handled by equipment placed at the local switching station.
While the dedicated circuit architecture of the circuit switched network was originally established for carrying an analog voice audio signal of a fixed bandwidth for the entire duration of the call, advances in technology enable digital data to be modulated on the twisted pair subscriber loop at very high data rates. For example, utilizing DSL technology, a telephony service provider can simultaneously provide both traditional telephony service (for one or more lines) as well as Internet access over a single subscriber loop by digitizing analog audio signals and utilizing a time division access scheme.
In one embodiment, often referred to as CBR, a time slot for supporting each telephony line remains permanently reserved on the subscriber loop regardless of whether such line is active (e.g. off-hook) or inactive (e.g on-hook) and a separate time slot for supporting packet switched data supports the provision of Internet access.
At the telephony service provider's switching station, both an Internet router and a telephone switch are coupled to the subscriber loop. During the time slots reserved for telephone service, the telephony switch communicates with a converter at the customer's premises. During the time slot reserved for Internet access, the Internet switch communicates with a DSL modem at the customer's premises.
More recently, telephone service has been implemented utilizing protocols known as voice over Internet Protocol (VoIP). Advances in the speed of data transmissions and Internet bandwidth have made it possible for telephone conversations to be communicated using the Internet's packet switched architecture with the overhead of the TCP/IP and UDP/IP protocols.
There exist several advantages of using VoIP to support one or more telephone lines over a DSL subscriber loop. First, the bandwidth of the subscriber loop is more efficiently allocated—dedicated time slots are not reserved for inactive telephone lines. Secondly, a combination of a VoIP-SS7 signaling gateway and a VoIP trunking gateway, located at any Internet addressable location, can replace a telephony switch at the telephony service provider's central office. Thirdly, calls placed to another VoIP line can be routed by a device commonly known as a “call agent” or “soft switch” as a peer-to-peer VoIP calls directly to the other endpoint across the Internet without use of a PSTN circuit.
However, a challenge with peer-to-peer VoIP telephony is that the telephony service provider's central office is completely bypassed making measuring of the call duration and billing for the call quite complicated. To facilitate the use of VoIP telephony with legacy billing systems, a device commonly known as a GR303 gateway has been developed.
A GR303 gateway, which is controlled by the telephony service provider, operates as a VoIP endpoint for all calls to and from a VoIP endpoint at the subscriber's premises. When a call is placed by the VoIP device at the subscriber's premises to a destination endpoint (either PSTN or VoIP), a call agent or soft switch directs the signaling to the GR303 gateway which immediately establishes a UDP/IP channel with the VoIP device and attempts to establish either a circuit switched connection over the PSTN or a VoIP connection over the Internet with the destination endpoint. After establishing the UDP/IP channel with the VoIP device, all further session signaling (such as busy tone and ring back tone) is provided in-band (e.g. as part of an audio signal that is digitized and compressed for transmission through the UDP/IP channel).
When a call is placed to the VoIP device at the subscriber's premises, the GR303 gateway receives the call and attempts to establish a UDP/IP channel with the VoIP device. After establishing the UDP/IP channel, all further session signaling (such as caller ID signals) are provide in band.
A problem associated with providing session signaling in the audio that is digitized and compressed for transmission over UDP/IP channels is that such signaling is only useful after the digitized and compressed audio is converted back to an analog or digital PSTN signal and is only interpretable by a traditional analog or digital PSTN device. When a multi-media terminal adapter (MTA) or other converter is used at the customer premises to emulate a PSTN analog or digital signal and the customer couples traditional PSTN equipment thereto, the PSTN equipment can readily interpret the session signaling.
However, if a VoIP telephone is used at the customer's premises, a traditional PSTN analog or digital signal is never generated and the in-band signaling remains unrecoverable. What is needed is a VoIP telephone with the capabilities of recognizing and interpreting signaling provided within digitized and compressed audio data over a UDP/IP channel with a remote gateway.